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Summary:ASTERISK-23765: RTP mishandling in chan_unistim
Reporter:Tamás Németh (nice0051)Labels:
Date Opened:2014-05-20 15:11:32Date Closed:2014-10-02 21:35:41
Priority:MajorRegression?Yes
Status:Closed/CompleteComponents:Channels/chan_unistim
Versions:12.2.0 Frequency of
Occurrence
Constant
Related
Issues:
Environment:LinuxAttachments:
Description:In Asterisk 11.x I used my i2001 and i2002 phones with rtp_method=1 but in asterisk 12.x there is no incoming voice on my unistim phones unless the partner is also an unistim phone. So I changed to rtp_method=3, which makes calls mutually audible, but RTP streams are somehow asymmetric: tcpdumping on the asterisk server I can see the RTP stream coming
from the unistim phone but no RTP stream goes from the server towards the phone! I assume that direction is directmedia or something like that. I tried call forwarding too, but it seems to be unable to handle this asymmetric half-direct, half -indirect RTP connection, and audio gets somehow confused and finally ceases to work.
Comments:By: Igor Goncharovsky (igorg) 2014-09-29 22:04:10.162-0500

If your phones works with rtp_method=1 with asterisk 11.x it MUST work with same setings on asterisk 12. Please try latest version, and attach console log with unistim debug enabled and tcpdump of unistim traf while call made.

By: Tamás Németh (nice0051) 2014-10-01 12:08:55.241-0500

OK, it works well in 12.5. Do I still have to send the debug data?

By: Igor Goncharovsky (igorg) 2014-10-02 21:34:32.414-0500

No it is seems to be fixed, I'll close this issue

By: Igor Goncharovsky (igorg) 2014-10-02 21:35:41.449-0500

Issue already fixed in released versions.